1. Field of the Invention
The present invention is directed to a device and method for controlling the quality of service a data communication network and more particularly to a network terminating unit for maintaining the quality of service of the data communication network at or below a level independent of network traffic.
2. Description of the Related Art
Communication network operators who provide traditional and enhanced telephony services (e.g., voice calls, FAX, voice mails), typically via the well known Public Switched Telephone Network (PSTN), are increasingly using data networks to convey their communication signals. One particular well known and popular data network is commonly referred to as the Internet. A data communication network is a communication network in which communication signals are conveyed by communication devices ( e.g., wireline telephones, wireless telephones, computers, modems, facsimile machines, video transmitters and receivers, terminal adapters) throughout the network in digital form.
The communication signals associated with a data communication network (or any other communication network) are conveyed in accordance with a protocol. The protocol represents a particular set of rules by which all or some of the communication devices within a data network (and other types of communication networks) initiate communication, convey information and terminate communication. Thus, all or some of the communication devices which are part of a data communication network should transmit and receive communication signals in accordance with a protocol. In many data networks, these communications signals, which represent some type of information (e.g., digital data, digitized voice, digitized video, facsimile data, protocol information) are structured as packets. Packets typically consist of header and/or trailer bits plus the user data to be conveyed. The header and/or trailer bits on each packet contains information required by data network protocols, such as source and destination addresses, control information, error checking and/or correction bits etc. The user information in a packet is often called its payload. Packets are transmitted through a packet network using a well known technique called packet switching.
There are many types of packet networks, including variations known as frame or cell networks, e.g., frame relay networks, Asynchronous Transfer Mode (ATM) networks etc. The term packet network refers to all types of data communication networks that transmit, receive, switch and otherwise process information between endpoints as discrete units, i.e., packets, frames, cells, etc.
In a packet switching data network, each packet is routed from point to point within the data communication network. The path taken by one packet representing part of a communication signal can be different from the path taken by other packets of that same communication signal. Thus, information can be represented by a communication signal which comprises at least one packet. One particular type of communication system which uses a data network to convey its communication signals is known as Packet Telephony.
Packet Telephony is the integration of speech compression and data networking technologies to provide traditional and enhanced telephony services over packet switched data networks rather than the PSTN or some other telephony system. Thus a packet telephony system comprises at least one underlying data network through which communication signals associated with the packet telephony system are conveyed. For example, if two users of the PSTN (Person A and Person B) are having a telephone conversation, the analog speech signals from A""s microphone are digitized by an A/D converter, typically at 8000 samples/second, 8 bits/sample, totaling 64 Kbits/second. The digital speech samples are then compressed to reduce the number of bits needed to represent them. The compression ratio is typically in the range of 8:1 to 10:1 yielding a bit rate in the range of 6400 bits/second to 8000 bits/sec. The compressor""s output is then formed into packets which are transmitted through a packet switched data communication network to the packet telephony system serving person B. When the packets are received by Person B""s system, the packets are de-packetized, i.e., the header and trailer bits are removed, and the remaining compressed information bits (i.e., compressed digitized voice) are sent to a decompressor. The decompressor output is connected to a D/A converter which drives Person B""s speaker. For a typical 2-party telephone call, packet telephony terminating equipment at each end simultaneously implement both the transmit and the receive functions.
FIG. 1 shows an exemplary configuration of a packet telephony communication system. For the sake of clarity, only two telephones (100, 116) are shown connected to an underlying data network 108 via packet telephony terminating equipment 104 and 112. In an actual packet telephony communication system there may be hundreds or even thousands of telephones and other communication devices connected to data communication network 108 via packet telephony terminating equipment (e.g., 104, 112). Also, an actual packet telephony communication system may use more than one data communication network to convey its communication signals.
Still referring to FIG. 1, telephones (100, 116) are connected to packet telephony terminating equipment (104, 112) via communication links 102 and 114. Communication links 102 and 114 as well as 106 and 110 can be any medium (e.g., twisted wire pairs, coaxial cables, fiber optic cables, air) through which communication signals are typically conveyed. Packet Telephony terminating equipment 104 and 112, which can be implemented with well known standard communication equipment such as gateways and routers, accept standard analog or constant bit rate digital voice signals (and other signals), encode these signals using voice compression techniques, and create packets from the resulting bit stream. The packets are forwarded to data communication network 108 which routes them to other packet telephony terminating equipment that extract the information bits from the packets, decompress the information bits, convert said bits to analog signals which are then sent to the telephone handsets (e.g., 100, 116) or other communication devices. Thus, for a 2-party packet telephone call as depicted in FIG. 1, the Packet Telephony network terminating equipment (104, 112) at each end of the call are simultaneously compressing, packetizing, decompressing and de-packetizing voice signals and other types of communication signals conveyed in a packet telephony system. A packet telephony call is defined as a telephone call between at least two users of a packet telephony system whereby the call is made in accordance with the protocols being followed by the packet telephony system; the packet telephony call includes voice calls, facsimile communications, voice mails and other services.
Prior to the use of packet switching in communication networks, many communication networks used a different scheme known as circuit switching. In contrast to packet switching, circuit switching allocates network resources to define a specific communication path or channel through which communication signals are to be conveyed between two points within the network. Circuit switching is widely employed in the design of telephony systems such as the well known POTS (Plain Old Telephone Service) networks or the PSTN in which a particular communication path, or channel or circuit is allocated specifically for particular users who wish to communicate with each other. Because of the manner in which circuit switching networks allocate their resources, circuit switched networks, such as the PSTN, are generally viewed as inefficient relative to packet switching networks particularly for sporadic or bursty communications.
Unlike circuit switched networks, data communication networks which use a packet switching scheme do not typically reserve resources for each active user; this increases the utilization efficiency of the data communication network infrastructure with bursty traffic, but makes the end-to-end performance highly dependent on the (usually uncontrollable) traffic patterns of all the users of the data communication network.
The quality of service provided by a packet telephony system (as perceived by users or operators of the packet telephony system) depends on the values of several well known network parameters of the underlying data communication network such as packet loss, bit errors, delay, delay variation (often called jitter) that affect the quality of a packet telephony call. As such, many data communication networks are capable of varying levels of quality of service. When packet telephony calls are conveyed over data communication networks together with varying data traffic, the quality of the calls suffers from breakups, dropouts or other unacceptable interruptions in the conversations as the parameter values vary outside acceptable ranges. Data communication networks can be engineered to provide acceptable (as defined by the users and/or network operators) quality of service for packet telephony calls under worst case loading situations. Such a design approach may, for example, be used for a packet telephony system that is intended to approximate the quality of service of the PSTN.
The quality of service of a communication system including a packet telephony system is defined by a set of values for one or more of the network parameters. For example, a network operator can define an acceptable quality of service for a particular packet telephony system as having a bit error rate of 10% averaged over any one second interval; a packet loss rate of 1 packet for every 10,000 packets transmitted and a propagation delay of 25 msec for any packet. Each parameter can also be characterized by a range of values. For example, an acceptable packet loss rate can be 1-5 packets loss for every 10,000 packets transmitted, an acceptable packet propagation delay can be 25-30 msec. Thus, for the last example, a packet loss rate of 1 or 2 or 3 or 4 packets is acceptable. The acceptable quality of service for another network may be defined by more or less network parameters or even only one network parameter. The set of network parameters and their values (or range of values) used to define an acceptable quality of service for any particular communication system (including a packet telephony system) can be devised by a network operator, the users of the system or both entities.
It is desirable for many network operators to use packet telephony systems to provide xe2x80x98off-brandxe2x80x99 or xe2x80x98second tierxe2x80x99 services defined as services whose quality and price are both lower than the PSTN; these operators may not wish to cannibalize their traditional profitable PSTN services by offering a new service with comparable quality but lower prices. The second tier or off-brand services should, of course, perform at acceptable levels, i.e., provide acceptable quality of service as defined by the network operators and/or users. However, given the difficulty of controlling the end-to-end quality of packet telephony calls due to the effects of unpredictable traffic loading within the underlying data communication network (i.e., network loading), operators are concerned that the quality of service of these second tier offerings can be quite acceptable (to users and/or network operators) during periods of low network load, and degrade to the expected lower quality levels as the loading increases. Alternatively, if the packet telephony system were engineered to provide the desired less-than-PSTN quality of service during periods of light traffic, heavy loading is likely to make the service totally unusable. The network load can be defined as the amount of users"" traffic that is being conveyed through a packet telephony system at any one time. The network load can also be defined as the percentage of the resources contained within the packet telephony system which are being used at any one time. The network load, thus, should reflect the capacity of the packet telephony system that is available to potential connected users. The variation in the quality of packet phone calls with network loading produces random changes in the performance of the packet telephony system as perceived by the subscribers (or network operators), leading to customer complaints and mismanaged user and operator expectations.
Therefore, there is a need for network operators to provide a packet telephony system that has an acceptable quality of service and which is independent of network loading. There is the further need for network operators to maintain the quality of service of the packet telephony system at a level that is below the quality of service of a traditional telephony system (e.g., PSTN) independent of the network load. There is yet a further need for network operators to provide a packet telephony system whereby the parameters of the underlying data networks can be maintained at certain values or range of values independent of network loading.
The present inventions provides a network terminating unit configured to receive communication signals associated with a data communication network capable of varying levels of quality of service. The network terminating unit comprises at least one module which can modify the received communication signals so as to keep the quality of service of the data communication network at or below a threshold level independent of network loading.